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Médias (1)
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DJ Dolores - Oslodum 2004 (includes (cc) sample of “Oslodum” by Gilberto Gil)
15 septembre 2011, par
Mis à jour : Septembre 2011
Langue : English
Type : Audio
Autres articles (69)
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Personnaliser en ajoutant son logo, sa bannière ou son image de fond
5 septembre 2013, parCertains thèmes prennent en compte trois éléments de personnalisation : l’ajout d’un logo ; l’ajout d’une bannière l’ajout d’une image de fond ;
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Ecrire une actualité
21 juin 2013, parPrésentez les changements dans votre MédiaSPIP ou les actualités de vos projets sur votre MédiaSPIP grâce à la rubrique actualités.
Dans le thème par défaut spipeo de MédiaSPIP, les actualités sont affichées en bas de la page principale sous les éditoriaux.
Vous pouvez personnaliser le formulaire de création d’une actualité.
Formulaire de création d’une actualité Dans le cas d’un document de type actualité, les champs proposés par défaut sont : Date de publication ( personnaliser la date de publication ) (...) -
Publier sur MédiaSpip
13 juin 2013Puis-je poster des contenus à partir d’une tablette Ipad ?
Oui, si votre Médiaspip installé est à la version 0.2 ou supérieure. Contacter au besoin l’administrateur de votre MédiaSpip pour le savoir
Sur d’autres sites (9775)
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How to choosing audio stream
5 septembre 2013, par user2696962I want to choose the first audio stream. but for some reason, it's not working. it's always choosing the last audio stream
what i've tried
ffmpeg -i test.avi -vf "movie=watermark_720.png [watermark]; [in][watermark] overlay=main_w-overlay_w-10:10 [out]" -map 0:0 -map 0:1 -y -ar 44100 -ac 1 -vcodec libx264 -b 555K -threads 0 test.2.flv
ffmpeg -i test.avi
ffmpeg -i test.avi
ffmpeg version 1.2.3 Copyright (c) 2000-2013 the FFmpeg developers
built on Sep 5 2013 03:04:34 with gcc 4.7 (Debian 4.7.1-2)
configuration: --enable-gpl --enable-libass --enable-libfaac --enable-libfdk-aac --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libspeex --enable-librtmp --enable-libtheora --enable-libvorbis --enable-libvpx --enable-x11grab --enable-libx264 --enable-nonfree --enable-version3
libavutil 52. 18.100 / 52. 18.100
libavcodec 54. 92.100 / 54. 92.100
libavformat 54. 63.104 / 54. 63.104
libavdevice 54. 3.103 / 54. 3.103
libavfilter 3. 42.103 / 3. 42.103
libswscale 2. 2.100 / 2. 2.100
libswresample 0. 17.102 / 0. 17.102
libpostproc 52. 2.100 / 52. 2.100
Input #0, avi, from 'test.avi':
Metadata:
encoder : VirtualDubMod 1.5.10.2 (build 2540/release)
Duration: 01:37:21.64, start: 0.000000, bitrate: 2018 kb/s
Stream #0:0: Video: mpeg4 (Advanced Simple Profile) (XVID / 0x44495658), yuv420p, 720x404 [SAR 1:1 DAR 180:101], 29.97 tbr, 29.97 tbn, 29.97 tbc
Stream #0:1: Audio: ac3 ([0] [0][0] / 0x2000), 48000 Hz, stereo, fltp, 160 kb/s
Stream #0:2: Audio: ac3 ([0] [0][0] / 0x2000), 48000 Hz, 5.1(side), fltp, 448 kb/stest.2.flv comes out with
Stream #0:2: Audio: ac3 ([0] [0][0] / 0x2000), 48000 Hz, 5.1(side), fltp, 448 kb/s
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FFMpeg - Merge multiple rtmp stream inputs to a single rtmp output
5 septembre 2013, par Paulo Miguel AlmeidaI'm trying to combine/merge two rtmp streams and then publish 'em to another stream
Ex. :
ffmpeg -i rtmp://ip:1935/live/micMyStream7 -i rtmp://ip:1935/live/MyStream7 -strict -2 -f flv rtmp://ip:1935/live/bcove7
The scenario is the following, I got a stream which comes from an user's microphone that
is the first one (micMyStream7) and I also got a stream from another user but this one has audio and video(MyStream7).As they are talking to each other when a user is speaking, the other one would only be listening to and vice versa.
My idea is to set up a third stream called (bcove) which would "merge" both of them so that I could have spectators who would only be listening to the entire conversation between them.
This is the log that ffmpeg printed although I couldn't recognize any message which helped me out.
paulo@paulo-desktop:~$ ffmpeg -re -i rtmp://ip:1935/live/micMyStream7 -i rtmp://ip:1935/live/MyStream7 -strict -2 -f flv rtmp://ip:1935/live/bcove7
ffmpeg version N-56029-g2ffead9 Copyright (c) 2000-2013 the FFmpeg developers
built on Sep 4 2013 11:05:57 with gcc 4.7 (Ubuntu/Linaro 4.7.3-1ubuntu1)
configuration:
libavutil 52. 43.100 / 52. 43.100
libavcodec 55. 31.100 / 55. 31.100
libavformat 55. 16.100 / 55. 16.100
libavdevice 55. 3.100 / 55. 3.100
libavfilter 3. 83.102 / 3. 83.102
libswscale 2. 5.100 / 2. 5.100
libswresample 0. 17.103 / 0. 17.103
Input #0, flv, from 'rtmp://ip:1935/live/micMyStream7':
Metadata:
author :
copyright :
description :
keywords :
rating :
title :
presetname : Medium Bandwidth (300 Kbps) - VP6
creationdate : Wed Sep 4 16:41:52 2013
:
videodevice : Built-in iSight
videokeyframe_frequency: 5
audiodevice : External microphone
audiochannels : 1
audioinputvolume: 75
Duration: N/A, start: 0.000000, bitrate: 253 kb/s
Stream #0:0: Video: vp6f, yuv420p, 320x240, 204 kb/s, 44.83 tbr, 1k tbn, 1k tbc
Stream #0:1: Audio: mp3, 22050 Hz, mono, s16p, 49 kb/s
Input #1, flv, from 'rtmp://ip:1935/live/MyStream7':
Metadata:
author :
copyright :
description :
keywords :
rating :
title :
presetname : Custom
creationdate : Wed Sep 4 12:02:24 2013
:
videodevice : FaceTime HD Camera (Built-in)
videokeyframe_frequency: 5
audiodevice : Internal microphone
audiochannels : 1
audioinputvolume: 75
Duration: N/A, start: 0.000000, bitrate: 253 kb/s
Stream #1:0: Video: vp6f, yuv420p, 320x240, 204 kb/s, 45.08 tbr, 1k tbn, 1k tbc
Stream #1:1: Audio: mp3, 22050 Hz, mono, s16p, 49 kb/s
Output #0, flv, to 'rtmp://ip:1935/live/bcove7':
Metadata:
author :
copyright :
description :
keywords :
rating :
title :
presetname : Medium Bandwidth (300 Kbps) - VP6
creationdate : Wed Sep 4 16:41:52 2013
:
videodevice : Built-in iSight
videokeyframe_frequency: 5
audiodevice : External microphone
audiochannels : 1
audioinputvolume: 75
encoder : Lavf55.16.100
Stream #0:0: Video: flv1 (flv) ([2][0][0][0] / 0x0002), yuv420p, 320x240, q=2-31, 200 kb/s, 1k tbn, 44.83 tbc
Stream #0:1: Audio: adpcm_swf ([1][0][0][0] / 0x0001), 22050 Hz, mono, s16, 88 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (vp6f -> flv)
Stream #0:1 -> #0:1 (mp3 -> adpcm_swf)
Press [q] to stop, [?] for help
[mp3 @ 0x3625ec0] overread, skip -9 enddists: -3 -300:14.44 bitrate= 224.0kbits/s
[mp3 @ 0x3625ec0] overread, skip -7 enddists: -3 -30:26.39 bitrate= 203.5kbits/sThanks in advance
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RED5 1.0.2 recorded FLV convert to Mobile/HTML5 format with FFMPEG av out of sync
14 août 2014, par Daew daweI have problem with recorded video in Red5 v1.0.2 (i had issues with previous v1, it does not recorded any video, with 1.0.2 it works). When I record flv I want to convert it to some mp4. But I have problem with settings, because every time there is some issue with quality/audio sync. Can u please help me how to convert with ffmpeg (in future automatic process on server).
Second problem is that in flash client buffer length is always 0, but in v0.8 it was filled and on end I waited until empty, here I’m not sure how long should I wait. I founded this url http://code.google.com/p/red5/issues/detail?id=312 where they said to wait until i get UnPublish.Success, but that event I got only after ns.close()
My flash client record settings is (FP10) :
video :
- resolution = 640x360
- fps = 30
- keyframeinterval = 15
- video quality = 90
- bandwidth = 0
audio :
- microphone codec = SPEEX
- encodeQuality = 9
-
silencelevel = 0
-
bufferTime = 15
recorded video parameters in VLC (translated from czech to english) :
video
- Codec : Flash Video (FLV1)
- Resolution : 640x360
- format : Planar 4:2:0 YUV
audio
- codec : Speex Audio (spx )
- frequency : 16000Hz
- bits per sample : 16
- data flow : 16 kb/s
FFMEPG info about video :
Metadata:
server : Red5 Server 1.0.2 Rev: 4616
creationdate : Mon Sep 02 23:17:08 CEST 2013
canSeekToEnd : true
Duration: 00:00:33.24, start: 0.000000, bitrate: 645 kb/s
Stream #0:0: Video: flv1, yuv420p, 640x360, 625 kb/s, 1k tbr, 1k tbn, 1k tbc
Stream #0:1: Audio: speex, 16000 Hz, mono, s16, 16 kb/sbsplayer showing 25fps - but I recorded 30fps, I dont understand this so much.
what I tried with ffmpeg (I’m ffmpeg newbie).
First I recorded 33sec long video
when I convert audio with command :
ffmpeg -i test.flv -ar 44100 -ab 160k -ac 1 output.mp3
, then the audio have only 30secI tried this commands, but no one with good solution
ffmpeg -i test.flv -vcodec mpeg4 -acodec libvo_aacenc output.mp4
ffmpeg -i test.flv -acodec libvo_aacenc -aq 200 outputsss.mp4
ffmpeg -i test.flv -c:v libvpx -c:a libvorbis output.webm // here is sound synced good - but sound have repeating silence lags (every 1-2s)really thank you for your help, I’m fighting with conversion many days :(